Return-Path: From: Andrei Emeltchenko To: linux-bluetooth@vger.kernel.org Subject: [PATCHv5 11/13] android/audio: Use resampler interface to resample SCO Date: Fri, 9 May 2014 12:19:53 +0300 Message-Id: <1399627195-471-11-git-send-email-Andrei.Emeltchenko.news@gmail.com> In-Reply-To: <1399627195-471-1-git-send-email-Andrei.Emeltchenko.news@gmail.com> References: <1399627195-471-1-git-send-email-Andrei.Emeltchenko.news@gmail.com> Sender: linux-bluetooth-owner@vger.kernel.org List-ID: From: Andrei Emeltchenko Resample Android audio from 44100 to 8000. --- android/hal-sco.c | 112 ++++++++++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 105 insertions(+), 7 deletions(-) diff --git a/android/hal-sco.c b/android/hal-sco.c index 09eea1c..1256144 100644 --- a/android/hal-sco.c +++ b/android/hal-sco.c @@ -27,6 +27,7 @@ #include #include +#include #include "../src/shared/util.h" #include "sco-msg.h" @@ -34,6 +35,7 @@ #include "hal-log.h" #define AUDIO_STREAM_DEFAULT_RATE 44100 +#define AUDIO_STREAM_SCO_RATE 8000 #define AUDIO_STREAM_DEFAULT_FORMAT AUDIO_FORMAT_PCM_16_BIT #define OUT_BUFFER_SIZE 2560 @@ -60,6 +62,10 @@ struct sco_stream_out { int fd; uint8_t *downmix_buf; + + struct resampler_itfe *resampler; + int16_t *resample_buf; + uint32_t resample_frame_num; }; struct sco_audio_dev { @@ -67,6 +73,22 @@ struct sco_audio_dev { struct sco_stream_out *out; }; +/* + * return the minimum frame numbers from resampling between BT stack's rate + * and audio flinger's. For output stream, 'output' shall be true, otherwise + * false for input streams at audio flinger side. + */ +static size_t get_resample_frame_num(uint32_t sco_rate, uint32_t rate, + size_t frame_num, bool output) +{ + size_t resample_frames_num = frame_num * sco_rate / rate + output; + + DBG("resampler: sco_rate %d frame_num %zd rate %d resample frames %zd", + sco_rate, frame_num, rate, resample_frames_num); + + return resample_frames_num; +} + /* Audio IPC functions */ static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len, @@ -258,6 +280,9 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, { struct sco_stream_out *out = (struct sco_stream_out *) stream; size_t frame_num = bytes / audio_stream_frame_size(&out->stream.common); + size_t output_frame_num = frame_num; + void *send_buf = out->downmix_buf; + size_t total; DBG("write to fd %d bytes %zu", out->fd, bytes); @@ -267,6 +292,34 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, } downmix_to_mono(out, buffer, frame_num); + + if (out->resampler) { + int ret; + + /* limit resampler's output within what resample buf can hold */ + output_frame_num = out->resample_frame_num; + + ret = out->resampler->resample_from_input(out->resampler, + send_buf, + &frame_num, + out->resample_buf, + &output_frame_num); + if (ret) { + error("Failed to resample frames: %zd input %zd (%s)", + frame_num, output_frame_num, strerror(ret)); + return -1; + } + + send_buf = out->resample_buf; + + DBG("Resampled: frame_num %zd, output_frame_num %zd", + frame_num, output_frame_num); + } + + total = output_frame_num * sizeof(int16_t) * 1; + + DBG("total %zd", total); + return bytes; } @@ -299,19 +352,18 @@ static size_t out_get_buffer_size(const struct audio_stream *stream) static uint32_t out_get_channels(const struct audio_stream *stream) { - DBG(""); + struct sco_stream_out *out = (struct sco_stream_out *) stream; - /* AudioFlinger can only provide stereo stream, so we return it here and - * later we'll downmix this to mono in case codec requires it - */ - return AUDIO_CHANNEL_OUT_STEREO; + DBG("channels num: %u", popcount(out->cfg.channels)); + + return out->cfg.channels; } static audio_format_t out_get_format(const struct audio_stream *stream) { struct sco_stream_out *out = (struct sco_stream_out *) stream; - DBG(""); + DBG("format: %u", out->cfg.format); return out->cfg.format; } @@ -401,6 +453,8 @@ static int audio_open_output_stream(struct audio_hw_device *dev, struct sco_audio_dev *adev = (struct sco_audio_dev *) dev; struct sco_stream_out *out; int fd = -1; + int chan_num, ret; + size_t resample_size; uint16_t mtu; DBG(""); @@ -433,8 +487,9 @@ static int audio_open_output_stream(struct audio_hw_device *dev, out->stream.write = out_write; out->stream.get_render_position = out_get_render_position; + /* Configuration for Android */ out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT; - out->cfg.channels = AUDIO_CHANNEL_OUT_MONO; + out->cfg.channels = AUDIO_CHANNEL_OUT_STEREO; out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE; out->cfg.frame_num = OUT_STREAM_FRAMES; out->cfg.mtu = mtu; @@ -446,11 +501,54 @@ static int audio_open_output_stream(struct audio_hw_device *dev, } DBG("size %zd", out_get_buffer_size(&out->stream.common)); + + /* Channel numbers for resampler */ + chan_num = 1; + + ret = create_resampler(out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num, + RESAMPLER_QUALITY_DEFAULT, NULL, + &out->resampler); + if (ret) { + error("Failed to create resampler (%s)", strerror(ret)); + goto failed; + } + + DBG("Created resampler: input rate [%d] output rate [%d] channels [%d]", + out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num); + + out->resample_frame_num = get_resample_frame_num(AUDIO_STREAM_SCO_RATE, + out->cfg.rate, + out->cfg.frame_num, 1); + + if (!out->resample_frame_num) { + error("frame num is too small to resample, discard it"); + goto failed; + } + + resample_size = sizeof(int16_t) * chan_num * out->resample_frame_num; + + out->resample_buf = malloc(resample_size); + if (!out->resample_buf) { + error("failed to allocate resample buffer for %u frames", + out->resample_frame_num); + goto failed; + } + + DBG("resampler: frame num %u buf size %zd bytes", + out->resample_frame_num, resample_size); + *stream_out = &out->stream; adev->out = out; out->fd = fd; return 0; +failed: + free(out->downmix_buf); + free(out); + stream_out = NULL; + adev->out = NULL; + + return ret; } static void audio_close_output_stream(struct audio_hw_device *dev, -- 1.8.3.2