[pulseaudio-discuss] [PATCH 52/56] bluetooth: Create I/O thread function for BlueZ 5 cards
João Paulo Rechi Vita
jprvita at gmail.com
Thu Jul 25 22:51:41 PDT 2013
On Thu, Jul 25, 2013 at 6:29 AM, Tanu Kaskinen
<tanu.kaskinen at linux.intel.com> wrote:
> On Fri, 2013-07-12 at 15:07 -0300, jprvita at gmail.com wrote:
>> From: João Paulo Rechi Vita <jprvita at openbossa.org>
>>
>> Create the thread function, the render and push functions for A2DP, the
>> process message function for communication between the I/O thread and
>> the main thread, and other helper functions related to them.
>> ---
>> src/modules/bluetooth/module-bluez5-device.c | 637 +++++++++++++++++++++++++++
>> 1 file changed, 637 insertions(+)
>>
>> diff --git a/src/modules/bluetooth/module-bluez5-device.c b/src/modules/bluetooth/module-bluez5-device.c
>> index d0971d5..9a5534c 100644
>> --- a/src/modules/bluetooth/module-bluez5-device.c
>> +++ b/src/modules/bluetooth/module-bluez5-device.c
>> @@ -24,19 +24,29 @@
>> #include <config.h>
>> #endif
>>
>> +#include <errno.h>
>> +
>> +#include <arpa/inet.h>
>> #include <sbc/sbc.h>
>>
>> +#include <pulse/rtclock.h>
>> +#include <pulse/timeval.h>
>> +
>> +#include <pulsecore/core-error.h>
>> #include <pulsecore/core-util.h>
>> #include <pulsecore/i18n.h>
>> #include <pulsecore/module.h>
>> #include <pulsecore/modargs.h>
>> #include <pulsecore/poll.h>
>> #include <pulsecore/rtpoll.h>
>> +#include <pulsecore/socket-util.h>
>> #include <pulsecore/thread.h>
>> #include <pulsecore/thread-mq.h>
>> +#include <pulsecore/time-smoother.h>
>>
>> #include "a2dp-codecs.h"
>> #include "bluez5-util.h"
>> +#include "rtp.h"
>>
>> #include "module-bluez5-device-symdef.h"
>>
>> @@ -46,11 +56,30 @@ PA_MODULE_VERSION(PACKAGE_VERSION);
>> PA_MODULE_LOAD_ONCE(false);
>> PA_MODULE_USAGE("path=<device object path>");
>>
>> +#define MAX_PLAYBACK_CATCH_UP_USEC (100 * PA_USEC_PER_MSEC)
>> +#define FIXED_LATENCY_PLAYBACK_A2DP (25 * PA_USEC_PER_MSEC)
>> +#define FIXED_LATENCY_RECORD_A2DP (25 * PA_USEC_PER_MSEC)
>> +
>> +#define BITPOOL_DEC_LIMIT 32
>> +#define BITPOOL_DEC_STEP 5
>> +
>> static const char* const valid_modargs[] = {
>> "path",
>> NULL
>> };
>>
>> +enum {
>> + BLUETOOTH_MESSAGE_IO_THREAD_FAILED,
>> + BLUETOOTH_MESSAGE_MAX
>> +};
>> +
>> +typedef struct bluetooth_msg {
>> + pa_msgobject parent;
>> + pa_card *card;
>> +} bluetooth_msg;
>> +PA_DEFINE_PRIVATE_CLASS(bluetooth_msg, pa_msgobject);
>> +#define BLUETOOTH_MSG(o) (bluetooth_msg_cast(o))
>> +
>> typedef struct sbc_info {
>> sbc_t sbc; /* Codec data */
>> bool sbc_initialized; /* Keep track if the encoder is initialized */
>> @@ -86,12 +115,19 @@ struct userdata {
>> pa_thread_mq thread_mq;
>> pa_rtpoll *rtpoll;
>> pa_rtpoll_item *rtpoll_item;
>> + bluetooth_msg *msg;
>>
>> int stream_fd;
>> + int stream_write_type;
>> size_t read_link_mtu;
>> size_t write_link_mtu;
>> size_t read_block_size;
>> size_t write_block_size;
>> + uint64_t read_index;
>> + uint64_t write_index;
>> + pa_usec_t started_at;
>> + pa_smoother *read_smoother;
>> + pa_memchunk write_memchunk;
>> pa_sample_spec sample_spec;
>> struct sbc_info sbc_info;
>> };
>> @@ -210,6 +246,320 @@ static void connect_ports(struct userdata *u, void *new_data, pa_direction_t dir
>> }
>> }
>>
>> +/* Run from IO thread */
>> +static void a2dp_prepare_buffer(struct userdata *u) {
>> + size_t min_buffer_size = PA_MAX(u->read_link_mtu, u->write_link_mtu);
>> +
>> + pa_assert(u);
>> +
>> + if (u->sbc_info.buffer_size >= min_buffer_size)
>> + return;
>> +
>> + u->sbc_info.buffer_size = 2 * min_buffer_size;
>> + pa_xfree(u->sbc_info.buffer);
>> + u->sbc_info.buffer = pa_xmalloc(u->sbc_info.buffer_size);
>> +}
>> +
>> +/* Run from IO thread */
>> +static int a2dp_process_render(struct userdata *u) {
>> + struct sbc_info *sbc_info;
>> + struct rtp_header *header;
>> + struct rtp_payload *payload;
>> + size_t nbytes;
>> + void *d;
>> + const void *p;
>> + size_t to_write, to_encode;
>> + unsigned frame_count;
>> + int ret = 0;
>> +
>> + pa_assert(u);
>> + pa_assert(u->profile == PROFILE_A2DP_SINK);
>> + pa_assert(u->sink);
>> +
>> + /* First, render some data */
>> + if (!u->write_memchunk.memblock)
>> + pa_sink_render_full(u->sink, u->write_block_size, &u->write_memchunk);
>> +
>> + pa_assert(u->write_memchunk.length == u->write_block_size);
>
> u->write_block_size may change due to bitpool reduction. If there was
> audio left in write_memchunk due to an earlier EAGAIN error, this
> assertion will crash. a2dp_process_render() needs to prepare for
> write_block_size changing between
>
>> +
>> + a2dp_prepare_buffer(u);
>> +
>> + sbc_info = &u->sbc_info;
>> + header = sbc_info->buffer;
>> + payload = (struct rtp_payload*) ((uint8_t*) sbc_info->buffer + sizeof(*header));
>> +
>> + frame_count = 0;
>> +
>> + /* Try to create a packet of the full MTU */
>> +
>> + p = (const uint8_t *) pa_memblock_acquire_chunk(&u->write_memchunk);
>> + to_encode = u->write_memchunk.length;
>> +
>> + d = (uint8_t*) sbc_info->buffer + sizeof(*header) + sizeof(*payload);
>> + to_write = sbc_info->buffer_size - sizeof(*header) - sizeof(*payload);
>
> This assumes that sbc_info->buffer_size is big enough to hold at least
> rtp_header and rtp_payload structs. buffer_size is determined by the
> MTU, and there seems to be no check that the MTU values are sensible.
>
>> +
>> + while (PA_LIKELY(to_encode > 0 && to_write > 0)) {
>> + ssize_t written;
>> + ssize_t encoded;
>> +
>> + encoded = sbc_encode(&sbc_info->sbc,
>> + p, to_encode,
>> + d, to_write,
>> + &written);
>> +
>> + if (PA_UNLIKELY(encoded <= 0)) {
>> + pa_log_error("SBC encoding error (%li)", (long) encoded);
>> + pa_memblock_release(u->write_memchunk.memblock);
>> + return -1;
>> + }
>> +
>> + pa_assert_fp((size_t) encoded <= to_encode);
>> + pa_assert_fp((size_t) encoded == sbc_info->codesize);
>> +
>> + pa_assert_fp((size_t) written <= to_write);
>> + pa_assert_fp((size_t) written == sbc_info->frame_length);
>
> These equality assertions make me nervous. I suggest adding a comment:
>
> /* These assertions are safe, because sbc_encode() encodes exactly one
> * input block to exactly one output block (documented). If the buffer
> * sizes are too small, sbc_encode() returns zero or an error (not
> * documented), which we catch above. */
> pa_assert_fp((size_t) encoded <= to_encode);
> pa_assert_fp((size_t) encoded == sbc_info->codesize);
>
> pa_assert_fp((size_t) written <= to_write);
> pa_assert_fp((size_t) written == sbc_info->frame_length);
>
>> +
>> + p = (const uint8_t*) p + encoded;
>> + to_encode -= encoded;
>> +
>> + d = (uint8_t*) d + written;
>> + to_write -= written;
>> +
>> + frame_count++;
>> + }
>> +
>> + pa_memblock_release(u->write_memchunk.memblock);
>> +
>> + pa_assert(to_encode == 0);
>> +
>> + PA_ONCE_BEGIN {
>> + pa_log_debug("Using SBC encoder implementation: %s", pa_strnull(sbc_get_implementation_info(&sbc_info->sbc)));
>> + } PA_ONCE_END;
>> +
>> + /* write it to the fifo */
>> + memset(sbc_info->buffer, 0, sizeof(*header) + sizeof(*payload));
>> + header->v = 2;
>> + header->pt = 1;
>> + header->sequence_number = htons(sbc_info->seq_num++);
>> + header->timestamp = htonl(u->write_index / pa_frame_size(&u->sample_spec));
>> + header->ssrc = htonl(1);
>> + payload->frame_count = frame_count;
>> +
>> + nbytes = (uint8_t*) d - (uint8_t*) sbc_info->buffer;
>> +
>> + for (;;) {
>> + ssize_t l;
>> +
>> + l = pa_write(u->stream_fd, sbc_info->buffer, nbytes, &u->stream_write_type);
>> +
>> + pa_assert(l != 0);
>> +
>> + if (l < 0) {
>> +
>> + if (errno == EINTR)
>> + /* Retry right away if we got interrupted */
>> + continue;
>> +
>> + else if (errno == EAGAIN)
>> + /* Hmm, apparently the socket was not writable, give up for now */
>> + break;
>> +
>> + pa_log_error("Failed to write data to socket: %s", pa_cstrerror(errno));
>> + ret = -1;
>> + break;
>> + }
>> +
>> + pa_assert((size_t) l <= nbytes);
>> +
>> + if ((size_t) l != nbytes) {
>> + pa_log_warn("Wrote memory block to socket only partially! %llu written, wanted to write %llu.",
>> + (unsigned long long) l,
>> + (unsigned long long) nbytes);
>> + ret = -1;
>> + break;
>> + }
>> +
>> + u->write_index += (uint64_t) u->write_memchunk.length;
>> + pa_memblock_unref(u->write_memchunk.memblock);
>> + pa_memchunk_reset(&u->write_memchunk);
>> +
>> + ret = 1;
>> +
>> + break;
>> + }
>> +
>> + return ret;
>> +}
>> +
>> +/* Run from IO thread */
>> +static int a2dp_process_push(struct userdata *u) {
>> + int ret = 0;
>> + pa_memchunk memchunk;
>> +
>> + pa_assert(u);
>> + pa_assert(u->profile == PROFILE_A2DP_SOURCE);
>> + pa_assert(u->source);
>> + pa_assert(u->read_smoother);
>> +
>> + memchunk.memblock = pa_memblock_new(u->core->mempool, u->read_block_size);
>> + memchunk.index = memchunk.length = 0;
>> +
>> + for (;;) {
>> + bool found_tstamp = false;
>> + pa_usec_t tstamp;
>> + struct sbc_info *sbc_info;
>> + struct rtp_header *header;
>> + struct rtp_payload *payload;
>> + const void *p;
>> + void *d;
>> + ssize_t l;
>> + size_t to_write, to_decode;
>> +
>> + a2dp_prepare_buffer(u);
>> +
>> + sbc_info = &u->sbc_info;
>> + header = sbc_info->buffer;
>> + payload = (struct rtp_payload*) ((uint8_t*) sbc_info->buffer + sizeof(*header));
>> +
>> + l = pa_read(u->stream_fd, sbc_info->buffer, sbc_info->buffer_size, &u->stream_write_type);
>
> The code assumes later that what we read here is exactly one complete
> rtp packet, nothing more, nothing less. I don't understand how such
> assumption can be done. I think we should parse the rtp header to find
> the packet boundary, and buffer any incomplete packets for later
> reading. Or perhaps recvmsg() should be used (I guess that's needed also
> for reading the timestamp, which is currently a TODO item).
>
>> +
>> + if (l <= 0) {
>> +
>> + if (l < 0 && errno == EINTR)
>> + /* Retry right away if we got interrupted */
>> + continue;
>> +
>> + else if (l < 0 && errno == EAGAIN)
>> + /* Hmm, apparently the socket was not readable, give up for now. */
>> + break;
>> +
>> + pa_log_error("Failed to read data from socket: %s", l < 0 ? pa_cstrerror(errno) : "EOF");
>> + ret = -1;
>> + break;
>> + }
>> +
>> + pa_assert((size_t) l <= sbc_info->buffer_size);
>> +
>> + u->read_index += (uint64_t) l;
>> +
>> + /* TODO: get timestamp from rtp */
>> + if (!found_tstamp) {
>> + /* pa_log_warn("Couldn't find SO_TIMESTAMP data in auxiliary recvmsg() data!"); */
>> + tstamp = pa_rtclock_now();
>> + }
>
> There are two kind of timestamps being talked about here: RTP timestamps
> and socket message timestamps. I don't know what we would use the socket
> message timestamps for, but I think supporting RTP timestamps would be
> very important, because it would allow us to detect dropped packets. We
> should generate silence as a substitute for the lost packets. Not
> detecting dropped packets means that our timing information gets messed
> up, making it impossible for module-loopback to work properly.
>
> Are the RTP timestamp semantics defined well enough so that we could
> know how much silence we need to generate? We seem to use the number of
> written samples in our outgoing packets, is that what we can expect also
> from incoming packets?
>
>> +
>> + pa_smoother_put(u->read_smoother, tstamp, pa_bytes_to_usec(u->read_index, &u->sample_spec));
>> + pa_smoother_resume(u->read_smoother, tstamp, true);
>> +
>> + p = (uint8_t*) sbc_info->buffer + sizeof(*header) + sizeof(*payload);
>> + to_decode = l - sizeof(*header) - sizeof(*payload);
>
> to_decode is unsigned, so this assumes that we read at least
> sizeof(*header) + sizeof(*payload) amount of data. That is not checked.
>
>> +
>> + d = pa_memblock_acquire(memchunk.memblock);
>> + to_write = memchunk.length = pa_memblock_get_length(memchunk.memblock);
>> +
>> + while (PA_LIKELY(to_decode > 0)) {
>> + size_t written;
>> + ssize_t decoded;
>> +
>> + decoded = sbc_decode(&sbc_info->sbc,
>> + p, to_decode,
>> + d, to_write,
>> + &written);
>> +
>> + if (PA_UNLIKELY(decoded <= 0)) {
>> + pa_log_error("SBC decoding error (%li)", (long) decoded);
>> + pa_memblock_release(memchunk.memblock);
>> + pa_memblock_unref(memchunk.memblock);
>> + return -1;
>> + }
>> +
>> + /* Reset frame length, it can be changed due to bitpool change */
>> + sbc_info->frame_length = sbc_get_frame_length(&sbc_info->sbc);
>
> read_block_size is derived from the frame length, so that needs to be
> updated too. And the source fixed latency depends on read_block_size.
> And so does the buffer length that was passed to sbc_decode, so it may
> happen that we passed too small buffer to sbc_decode().
>
>> +
>> + pa_assert_fp((size_t) decoded <= to_decode);
>> + pa_assert_fp((size_t) decoded == sbc_info->frame_length);
>> +
>> + pa_assert_fp((size_t) written == sbc_info->codesize);
>
> This will crash if the output buffer was too small (an undocumented
> feature of sbc_decode is that it truncates the output if the given
> buffer is too small, instead of failing like sbc_encode() does).
> Removing this assertion isn't the right fix, though, the right fix is to
> ensure that we have big enough output buffer.
>
>> +
>> + p = (const uint8_t*) p + decoded;
>> + to_decode -= decoded;
>> +
>> + d = (uint8_t*) d + written;
>> + to_write -= written;
>> + }
>> +
>> + memchunk.length -= to_write;
>> +
>> + pa_memblock_release(memchunk.memblock);
>> +
>> + pa_source_post(u->source, &memchunk);
>> +
>> + ret = l;
>> + break;
>> + }
>> +
>> + pa_memblock_unref(memchunk.memblock);
>> +
>> + return ret;
>> +}
>> +
>> +/* Run from I/O thread */
>> +static void a2dp_set_bitpool(struct userdata *u, uint8_t bitpool) {
>> + struct sbc_info *sbc_info;
>> +
>> + pa_assert(u);
>> +
>> + sbc_info = &u->sbc_info;
>> +
>> + if (sbc_info->sbc.bitpool == bitpool)
>> + return;
>> +
>> + if (bitpool > sbc_info->max_bitpool)
>> + bitpool = sbc_info->max_bitpool;
>> + else if (bitpool < sbc_info->min_bitpool)
>> + bitpool = sbc_info->min_bitpool;
>> +
>> + sbc_info->sbc.bitpool = bitpool;
>> +
>> + sbc_info->codesize = sbc_get_codesize(&sbc_info->sbc);
>> + sbc_info->frame_length = sbc_get_frame_length(&sbc_info->sbc);
>> +
>> + pa_log_debug("Bitpool has changed to %u", sbc_info->sbc.bitpool);
>
> Could you calculate the resulting SBC bitrate and add it to the log
> message? I think that would be nice to see in the log, because the
> bitpool value doesn't really tell anything meaningful to the vast
> majority of people (including me). If you do that, log the bitrate also
> in transport_config() and when the bitpool changes in the a2dp_source
> mode.
>
>> +
>> + u->read_block_size =
>> + (u->read_link_mtu - sizeof(struct rtp_header) - sizeof(struct rtp_payload))
>> + / sbc_info->frame_length * sbc_info->codesize;
>> +
>> + u->write_block_size =
>> + (u->write_link_mtu - sizeof(struct rtp_header) - sizeof(struct rtp_payload))
>> + / sbc_info->frame_length * sbc_info->codesize;
>> +
>> + pa_sink_set_max_request_within_thread(u->sink, u->write_block_size);
>> + pa_sink_set_fixed_latency_within_thread(u->sink,
>> + FIXED_LATENCY_PLAYBACK_A2DP + pa_bytes_to_usec(u->write_block_size, &u->sample_spec));
>> +}
>> +
>> +/* Run from I/O thread */
>> +static void a2dp_reduce_bitpool(struct userdata *u) {
>> + struct sbc_info *sbc_info;
>> + uint8_t bitpool;
>> +
>> + pa_assert(u);
>> +
>> + sbc_info = &u->sbc_info;
>> +
>> + /* Check if bitpool is already at its limit */
>> + if (sbc_info->sbc.bitpool <= BITPOOL_DEC_LIMIT)
>> + return;
>> +
>> + bitpool = sbc_info->sbc.bitpool - BITPOOL_DEC_STEP;
>> +
>> + if (bitpool < BITPOOL_DEC_LIMIT)
>> + bitpool = BITPOOL_DEC_LIMIT;
>> +
>> + a2dp_set_bitpool(u, bitpool);
>> +}
>> +
>> static void teardown_stream(struct userdata *u) {
>> if (u->rtpoll_item) {
>> pa_rtpoll_item_free(u->rtpoll_item);
>> @@ -221,6 +571,16 @@ static void teardown_stream(struct userdata *u) {
>> u->stream_fd = -1;
>> }
>>
>> + if (u->read_smoother) {
>> + pa_smoother_free(u->read_smoother);
>> + u->read_smoother = NULL;
>> + }
>> +
>> + if (u->write_memchunk.memblock) {
>> + pa_memblock_unref(u->write_memchunk.memblock);
>> + pa_memchunk_reset(&u->write_memchunk);
>> + }
>> +
>> pa_log_debug("Audio stream torn down");
>> }
>>
>> @@ -258,6 +618,62 @@ static void transport_release(struct userdata *u) {
>> teardown_stream(u);
>> }
>>
>> +/* Run from I/O thread */
>> +static void transport_config_mtu(struct userdata *u) {
>> + u->read_block_size =
>> + (u->read_link_mtu - sizeof(struct rtp_header) - sizeof(struct rtp_payload))
>> + / u->sbc_info.frame_length * u->sbc_info.codesize;
>> +
>> + u->write_block_size =
>> + (u->write_link_mtu - sizeof(struct rtp_header) - sizeof(struct rtp_payload))
>> + / u->sbc_info.frame_length * u->sbc_info.codesize;
>> +
>> + if (u->sink) {
>> + pa_sink_set_max_request_within_thread(u->sink, u->write_block_size);
>> + pa_sink_set_fixed_latency_within_thread(u->sink,
>> + FIXED_LATENCY_PLAYBACK_A2DP +
>> + pa_bytes_to_usec(u->write_block_size, &u->sample_spec));
>> + }
>> +
>> + if (u->source)
>> + pa_source_set_fixed_latency_within_thread(u->source,
>> + FIXED_LATENCY_RECORD_A2DP +
>> + pa_bytes_to_usec(u->read_block_size, &u->sample_spec));
>> +}
>> +
>> +/* Run from I/O thread */
>> +static void setup_stream(struct userdata *u) {
>> + struct pollfd *pollfd;
>> + int one;
>> +
>> + pa_log_info("Transport %s resuming", u->transport->path);
>> +
>> + transport_config_mtu(u);
>> +
>> + pa_make_fd_nonblock(u->stream_fd);
>> + pa_make_socket_low_delay(u->stream_fd);
>> +
>> + one = 1;
>> + if (setsockopt(u->stream_fd, SOL_SOCKET, SO_TIMESTAMP, &one, sizeof(one)) < 0)
>> + pa_log_warn("Failed to enable SO_TIMESTAMP: %s", pa_cstrerror(errno));
>> +
>> + pa_log_debug("Stream properly set up, we're ready to roll!");
>> +
>> + if (u->profile == PROFILE_A2DP_SINK)
>> + a2dp_set_bitpool(u, u->sbc_info.max_bitpool);
>> +
>> + u->rtpoll_item = pa_rtpoll_item_new(u->rtpoll, PA_RTPOLL_NEVER, 1);
>> + pollfd = pa_rtpoll_item_get_pollfd(u->rtpoll_item, NULL);
>> + pollfd->fd = u->stream_fd;
>> + pollfd->events = pollfd->revents = 0;
>> +
>> + u->read_index = u->write_index = 0;
>> + u->started_at = 0;
>> +
>> + if (u->source)
>> + u->read_smoother = pa_smoother_new(PA_USEC_PER_SEC, 2*PA_USEC_PER_SEC, true, true, 10, pa_rtclock_now(), true);
>> +}
>> +
>> /* Run from main thread */
>> static int add_source(struct userdata *u) {
>> pa_source_new_data data;
>> @@ -496,6 +912,194 @@ static int init_profile(struct userdata *u) {
>>
>> /* I/O thread function */
>> static void thread_func(void *userdata) {
>> + struct userdata *u = userdata;
>> + unsigned do_write = 0;
>> + unsigned pending_read_bytes = 0;
>> + bool writable = false;
>> +
>> + pa_assert(u);
>> + pa_assert(u->transport);
>> +
>> + pa_log_debug("IO Thread starting up");
>> +
>> + if (u->core->realtime_scheduling)
>> + pa_make_realtime(u->core->realtime_priority);
>> +
>> + pa_thread_mq_install(&u->thread_mq);
>> +
>> + /* Setup the stream only if the transport was already acquired */
>> + if (u->transport_acquired)
>> + setup_stream(u);
>> +
>> + for (;;) {
>> + struct pollfd *pollfd;
>> + int ret;
>> + bool disable_timer = true;
>> +
>> + pollfd = u->rtpoll_item ? pa_rtpoll_item_get_pollfd(u->rtpoll_item, NULL) : NULL;
>> +
>> + if (u->source && PA_SOURCE_IS_LINKED(u->source->thread_info.state)) {
>> +
>> + /* We should send two blocks to the device before we expect
>> + * a response. */
>
> What response? I don't understand what this logic is trying to achieve.
>
>> +
>> + if (u->write_index == 0 && u->read_index <= 0)
>
> Using == 0 with write_index and <= 0 with read_index looks odd.
> read_index isn't any more suspectible to be negative than write_index
> (both have unsigned type anyway).
>
>> + do_write = 2;
>> +
>> + if (pollfd && (pollfd->revents & POLLIN)) {
>> + int n_read;
>> +
>> + n_read = a2dp_process_push(u);
>> +
>> + if (n_read < 0)
>> + goto io_fail;
>> +
>> + /* We just read something, so we are supposed to write something, too */
>> + pending_read_bytes += n_read;
>> + do_write += pending_read_bytes / u->write_block_size;
>> + pending_read_bytes = pending_read_bytes % u->write_block_size;
>
> This is not relevant for A2DP. Although this doesn't actively do any
> harm in the A2DP mode either, for readability I think it would be a good
> idea to run this code only when it's actually useful.
>
>> + }
>> + }
>> +
>> + if (u->sink && PA_SINK_IS_LINKED(u->sink->thread_info.state)) {
>> +
>> + if (PA_UNLIKELY(u->sink->thread_info.rewind_requested))
>> + pa_sink_process_rewind(u->sink, 0);
>> +
>> + if (pollfd) {
>> + if (pollfd->revents & POLLOUT)
>> + writable = true;
>> +
>> + if ((!u->source || !PA_SOURCE_IS_LINKED(u->source->thread_info.state)) && do_write <= 0 && writable) {
>> + pa_usec_t time_passed;
>> + pa_usec_t audio_sent;
>> +
>> + /* Hmm, there is no input stream we could synchronize
>> + * to. So let's do things by time */
>> +
>> + time_passed = pa_rtclock_now() - u->started_at;
>
> u->started_at is 0 in the beginning, and initialized to a sensible value
> only at the time of the first write attempt (BTW, I think it should be
> initialized only if the write actually succeeds). So in the first round
> time_passed will be very large, when it should be zero.
>
> This doesn't cause practical problems, because skipping is triggered
> only if write_index is greater than zero, but I think the code would be
> clearer if time_passed would be assigned zero here.
>
>> + audio_sent = pa_bytes_to_usec(u->write_index, &u->sample_spec);
>> +
>> + if (audio_sent <= time_passed) {
>> + pa_usec_t audio_to_send = time_passed - audio_sent;
>> +
>> + /* Never try to catch up for more than 100ms */
>> + if (u->write_index > 0 && audio_to_send > MAX_PLAYBACK_CATCH_UP_USEC) {
>
> Here the "write_index > 0" check wouldn't be needed if time_passed would
> always have a sane value.
>
>> + pa_usec_t skip_usec;
>> + uint64_t skip_bytes;
>> +
>> + skip_usec = audio_to_send - MAX_PLAYBACK_CATCH_UP_USEC;
>> + skip_bytes = pa_usec_to_bytes(skip_usec, &u->sample_spec);
>> +
>> + if (skip_bytes > 0) {
>> + pa_memchunk tmp;
>> +
>> + pa_log_warn("Skipping %llu us (= %llu bytes) in audio stream",
>> + (unsigned long long) skip_usec,
>> + (unsigned long long) skip_bytes);
>> +
>> + pa_sink_render_full(u->sink, skip_bytes, &tmp);
>
> write_memchunk may contain buffered audio, which should be consumed
> first. I propose that the amount of audio to skip is saved in userdata,
> and handled within a2dp_process_render(), because fiddling with
> write_memchunk should ideally be done only in a2dp_process_render().
>
>> + pa_memblock_unref(tmp.memblock);
>> + u->write_index += skip_bytes;
>> +
>> + if (u->profile == PROFILE_A2DP_SINK)
>> + a2dp_reduce_bitpool(u);
>> + }
>> + }
>> +
>> + do_write = 1;
>> + pending_read_bytes = 0;
>> + }
>> + }
>> +
>> + if (writable && do_write > 0) {
>> + int n_written;
>> +
>> + if (u->write_index <= 0)
>> + u->started_at = pa_rtclock_now();
>> +
>> + if ((n_written = a2dp_process_render(u)) < 0)
>> + goto io_fail;
>> +
>> + if (n_written == 0)
>> + pa_log("Broken kernel: we got EAGAIN on write() after POLLOUT!");
>> +
>> + do_write -= n_written;
>> + writable = false;
>> + }
>> +
>> + if ((!u->source || !PA_SOURCE_IS_LINKED(u->source->thread_info.state)) && do_write <= 0) {
>
> We should enable the timer only if we have already successfully written
> something, so this condition should be added: "&& u->write_index > 0".
>
>> + pa_usec_t sleep_for;
>> + pa_usec_t time_passed, next_write_at;
>> +
>> + if (writable) {
>> + /* Hmm, there is no input stream we could synchronize
>> + * to. So let's estimate when we need to wake up the latest */
>> + time_passed = pa_rtclock_now() - u->started_at;
>> + next_write_at = pa_bytes_to_usec(u->write_index, &u->sample_spec);
>> + sleep_for = time_passed < next_write_at ? next_write_at - time_passed : 0;
>> + /* pa_log("Sleeping for %lu; time passed %lu, next write at %lu", (unsigned long) sleep_for, (unsigned long) time_passed, (unsigned long)next_write_at); */
>> + } else
>> + /* drop stream every 500 ms */
>> + sleep_for = PA_USEC_PER_MSEC * 500;
>
> I don't think this works. I guess "drop stream" refers to the skipping
> functionality, but skipping is only done if the socket is writable. If
> the 500 ms timer fires, it means that the socket is not writable
> (otherwise we would have woken up due to a POLLOUT event). If dropping
> data every 500 ms is something that we want to do, the skipping code
> needs to be modified so that it is run also if the socket is not
> writable.
>
> Phew, this one took long.
>
Yes, and this code is very complex and sensible, thanks a lot for such
a detailed review. But because of all the pending patches in this
series and the other ~15 to come that implements HSP/HFP support, I
suggest that we merge this piece of code how it is atm, since it has
been tested for a long time (it's basically the same code from BlueZ 4
support), and take care of the comments on this patch and 53/56 after
the release, what do you think?
--
João Paulo Rechi Vita
http://about.me/jprvita
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